A modern high-quality audio system has excellent specifications and sounds almost perfect. Almost perfect, but not quite. There is one very important attribute missing in audio systems—the attribute we call “presence”. This article discusses an alternative power amplifier design with sound that often lacks in conventional amplifiers.

Even the best commercially available audio systems lack real presence–while the sound can be crystal clear, you would never mistake the recorded voices for real voices, or the recorded piano for a real piano. The human ear immediately knows the difference.

As listeners, even as audiophile listeners, we don’t fuss about this lack of presence because we have come to accept that what we hear from a modern audio system is as good as it gets. Yet this just isn’t true, and it doesn’t have to be accepted.

The lack of presence occurs almost entirely as a result of distortions inherent in the fundamental design of all commercial power amplifiers. Have you noticed how much clearer headphones sound? It's due to the fact that they are driven by low-powered amplifiers.

"Block diagram of conventional class B amplifier with the two halves of a complementary output stage represented by sub-amplifiers X and Y. " From Peter Blomley's 1971 article (PDF).

In this article, I demonstrate that there is an alternative power amplifier design that suffers virtually no distortion, and provides a sound which has the presence so lacking in conventional amplifiers. This amplifier was designed over forty years ago, and yet despite its superb fidelity, it has never seen commercial production.

I will introduce the idea of “rogue frequencies” and their effect on our listening experience. I will then go on to show how this original and truly suburb amplifier successfully minimizes distortions and does not generate disturbing rogue frequencies.

This design is so effective and the output so pure that it creates an audio presence that is quite impossible to ignore.

To understand why commercial amplifiers produce a sound which is very good but which lacks presence, I will begin by discussing the sensitivity of the human ear. Then I will examine types of distortion and how these affect what we actually hear.

Finally, as a potential add-on to the original design, I will discuss distortion caused by “clipping” and how to reduce its harshness.

A Brief Anecdote about a Gramophone

Recently, I went to visit an enthusiast friend who wanted to show me a genuine 1890’s wind-up gramophone with a real thorn needle and a real “His Master’s Voice” horn that he had just bought.

His Master's Voice, painted 1898 by Francis Baurraud.

With the purchase of the gramophone came some very old 78rpm records made directly from the original wax master—the huge horn of the recording gramophone was placed close to the orchestra and the needle cut a spiral groove, using only sound energy, in a platter covered with a thin layer of wax. There was an original record in good condition of Caruso in full glorious voice! There were also some orchestral and choir records.

These were all made before the triode was invented so no electronics were used in their manufacture. To humor my friend I agreed to listen, expecting an unpleasantly distorted sound. As I suspected, there was hiss and there were crackles, but the music! Pure and clear, you could hear each and every instrument and voice completely separately and beautifully, even with a choir and a mono recording source. Caruso really does deserve his reputation. It sounded ALIVE and PRESENT, even though it was made in 1902.

So why do modern amplifiers, with all the remarkable improvements we have achieved in electronic technology, lack this essential quality of presence?

How the Human Ear Perceives Harmonic Distortion

The human ear is extraordinarily sensitive. Under ideal conditions, the ear can hear sounds from the eardrum moving by as little as the diameter of a hydrogen atom (~10-10 m). Curiously, while the human ear can be very sensitive to some things, it isn't very sensitive to other things. For example, to notice a change in volume, the power has to be doubled (3db).

So if you were to increase, say, the 2nd harmonic by 5%, it would be an extraordinary person who noticed. Thus, genuine pure harmonic distortion below about 10% is pretty undetectable and irrelevant for even the best hi-fi. However, harmonic distortion is easy for the engineer to measure to great levels of accuracy and down to very low levels—so it gets talked about a lot, even though it really doesn’t matter much in the long run!

Consider how you identify your mother’s voice instantly, even over the lo-fi telephone. It's done by harmonic content, and the human ear is very deeply tuned to harmonic content: "Have you got a cold, Mum!?" can be asked after just one sentence from her.

This shows us the introduction of extra harmonics is very audible indeed (as little as 0.01% is easily detectable as a different type of sound).

Intermodulation Distortion

Any two frequencies passed through a non-linear amplifier will produce the sum frequency and the difference frequency in addition to the original frequencies. The amplitude of these additional frequencies (“rogue frequencies”) is related to the amount of non-linearity. This is intermodulation distortion and it is very difficult to measure, especially at the extremely low levels that still remain significant to the human ear. These additional unwanted rogue frequencies are off-key on our standard musical scale, and even tiny amounts make the music sound “muddy”.

Music or voice consists of hundreds of superimposed frequencies at any given moment (as described by the mathematician Joseph Fourier). As this collection of frequencies is passed through an amplifier, additional small-amplitude "rogue frequencies" are added to the original signal, and the human ear is very sensitive to this additional frequency content, and immediately identifies that the sound is not real. This is a significant reason why you never confuse, say, voices on the TV or radio with real visitors even when you are in another room.

Even very good conventional amplifiers are not mistaken for the real thing! Transducers (such as a needle on a record or a loudspeaker) are usually reasonably linear (certainly the audiophile versions) so they do not introduce many rogue frequencies although their harmonic distortion (through resonances, etc.) can be quite large. The weakness in a hi-fi system, no matter its numerically apparent superb specifications, is usually only the amplifier.

Crossover Distortion

Amplifier analysis shows that a Class B amplifier has a no-feedback distortion of about 33% and a Class A amplifier has a no-feedback distortion of about 8% and sound better than Class B. Most of the Class B distortion comes from the use of the output transistors as rectifiers to separate the plus and minus halves of the signal as well as then amplifying those halves separately thereafter (“push-pull”).

When a power transistor is driven below a collector current of about 15mA the amplification falls dramatically. If one could prevent the current in the output power-transistors from ever going below about 15mA and into this non-linear region, it would considerably improve things. This change of amplification causes the crossover distortion characteristic of class-B amplifiers.

Note that this crossover distortion should not be confused with the audio distortion, often also referred to as crossover distortion, which arises when audio signals are separated into frequency bands, as in loudspeaker circuits to feed the appropriate frequency range to each discrete driver unit.

Transient Intermodulation Distortion

Modern amplifier distortion is controlled by negative feedback, which reduces the distortion in proportion to the feedback. Amplification is cheaply available so the apparent non-linearity can be reduced to arbitrarily low levels by sufficient feedback.

But the feedback signal takes time to get through the amplifier and back to the input negatively to quash the distortion. So when sudden changes (transients) occur there is a period during which the naked amplifier is exposed to the world, and the non-linearity adds intermodulation rogue signals to the original, which are not entirely canceled by the feedback. This is transient intermodulation distortion. What you need is an amplifier sensibly without distortion before applying feedback The distortion of a naked class-B amplifier is so bad that most analysis seems to only consider the with-feedback distortion.

One of the reasons that modern amplifiers sound better than their older counterparts (using essentially the same class A or B circuits as always) is the increase in speed of the components. Multi-gigahertz discrete components are freely available, and even cheap power transistors have an ft of many MHz. This means that the feedback time is now very short indeed.

Clipping Distortion and How to Soften It

Transistor amplifiers driven into saturation sound horrible because the tops of the waveform are very sharply clipped off, leading to square corners and a huge explosion of unpleasant harmonics. I find myself waiting to wince when a conventional amplifier is driven hard.

On the other hand, near their clipping point, valves have quite a soft non-linear characteristic, resulting in a rounded squashed sine-wave which contain fewer spurious harmonics and sounds much better than the square-clipped sine-wave of a transistor amplifier driven hard.

The complete Blomley Amplifier with complementary output and clipping protection. Click to enlarge.

It measures the combination of voltage (RP3) and current (= voltage across RP2) in the output transistors and when the combination of these two voltages exceeds about 0.6V BE it turns on TRP and removes the drive to the outputs.

Note that the 0.6V is nominal and some current starts to flow when Vbe exceeds ~0.45V so this is a “soft” turn-off. It has two advantages:

It makes the amplifier “clip” softly, very similar to valve designs, which makes the sound very forgiving and prevents the “cringes”.

It protects the output transistors from most abuse.

Peter Blomley's New Approach to Class B Amplifier Design

In the February and March 1971 editions of Wireless World, Peter Blomley published the revolutionary and very densely concentrated article “New Approach to Class B Amplifier Design” (PDF) in two parts (patented by Plessey, No.53916.69, though this patent has long expired).

The very clever bit of the amplifier he describes is that Blomley split the incoming signal into top and bottom halves before applying the separate signals to the output transistors. Then he was easily able to design the output transistors to work only in their linear region (above a collector current of ~15mA).

Additionally, he made the observation that with voltage signals, diodes are very non-linear, but if you use a current source, the diode is so close to the theoretical ideal that one can really call it perfect (109 difference between forward and backward current in cheap diodes).

As shown in the schematic above, he used a constant-current source (Tr6) and had a varying current sink (Tr3). The current-difference drives diodes, which are actually transistors used as diodes, (Tr4 & Tr5) to rectify the current. By using very high-frequency transistors here the transition from the “top” signal to the “bottom” signal was so fast that it was way beyond 100kHz.

"New approach to class B amplifier in which SUb-amplifiers are biased above non-linear region and fed with uni-directional signals produced by the diodes. This effectively transfers signal splitting from the sub-amplifiers to a separate part of the circuit." From Peter Blomley (PDF).

The end result of this difficult-to-understand circuitry (we are used to voltage circuits) is a Class B amplifier that has a distortion lower than 0.1% with no feedback at all. And on an oscilloscope, there is no discernible crossover distortion with no feedback.

After a little feedback is applied, there is unmeasurable intermodulation distortion, transient intermodulation distortion, and harmonic distortion. The resultant output of this amplifier is so clear that a recorded voice can easily be mistaken for a live person. Peter Blomley’s amplifier is a Class B amplifier with much better than Class A performance.

And yet Peter Blomley and his amplifier have gone virtually unrecognized in the audio world for more than 40 years. I suggest two reasons for this. First, his design was so original and so unexpected that few people understood it or took it seriously. Second, Blomley never put his design into commercial production because Plessey held the patent, so even fewer people were able to listen to it or review its performance.

Most of the audio hobbyists who constructed their own Blomley amplifier modified the design and in doing so introduced distortions. I suggest you build the original design (with perhaps just the minor modifications afforded by modern components) and listen to it. This will give you a reference sound to check any further modifications with which you might like to experiment.

Unfortunately, in ignoring the Blomley design for so long, the audio world has deprived itself of a fundamentally better amplifier. We have instead put all our efforts over the last forty years in trying to mitigate what we thought were unavoidable inherent characteristics of electronic amplifiers, particularly Class B amplifiers. The boldness of Peter Blomley as a young engineer was to question how unavoidable these characteristics really were, and to then set about designing them out of his amplifier.

Today, superb high-voltage, high-speed transistors are available which makes the Blomley amplifier even better than his 1971 version.

The original amplifier design was for a 30W amplifier with a 60V power-rail, and because of the purity, this is more than adequate for normal home use. In 1971, 100V small-signal transistors were rare, but this is no longer so and an 80V power-rail can now be used, with different transistors, increasing the power to 50W. However, high sound volumes are not needed as the sound is so exceptionally clean. The huge headroom provided with most amplifiers is there so you can play them at high volume and bury the crossover-caused intermodulation distortion in the high sound-level (quite sad really).

Conclusion

40 years ago I fell in love with the clarity and purity of the sound from the Blomley amplifier, but it took a long time to understand the circuit and to appreciate the brilliance of Peter Blomley. Now, I have built several of these amplifiers and, provided I stick to the original Blomley design, they all sounded better than superb.

Human response, including our own, is often difficult to explain, but I have found that with a Blomley amplifier ordinary people find themselves wanting to listen to music much more than they do with a conventional top-end amplifier design. They don’t understand why, they just end up listening to more music, more often—surely the ultimate test. Listening to a Blomley amplifier is addictive. I have certainly found it so, as have many others fortunate enough to have experienced this extraordinary amplifier.

In fact, it is difficult to use it for background music; people tend to stop talking and start listening to the music. Its presence is compelling.

In his article, Peter Blomley expressed the very 1970s thought that states “The performance of an amplifier of this caliber is, in my opinion, wasted in a conventional audio set-up.” I built my first Blomley, and immediately realized that I could not agree with the sentiment he had expressed. My mother, who was garrulous in the extreme, sat through the whole of the “Pirates of Penzance” without saying a word! That was the 1970s and the other components in an audio system have come a long way since then.

The choice of a Blomley amplifier is now certainly warranted and is the best way to benefit fully from the technical advances made in all the other components of an audio system.

Notes on the Circuit

In most of the amplifiers I have built I have used a quasi-complementary output transistor arrangement but nowadays matched complementary power transistors are easily available. It really doesn't seem to matter!

Great care must be taken to physically separate the input from the output to prevent high-frequency feedback. This amplifier is quick enough to use at RF frequencies.

I have the LTSpice file if anyone is interested in playing with it.

References

“New Approach to Class B Amplifier Design” by Peter Blomley in Wireless World February and March 1971

Because this is a completely normal (from the point of view of the output transistors) class B amplifier, the dissipation is only from working.
I use an idle-current of 15-20 mA so the idle dissipation is ~1,2W. Of course at full power it gets hot as expected.

This is the most unfussy amplifier I have come across. In one application I used a few 1% resistors and eliminated the output idle-current adjustment potentiometers and all adjustment in a production circuit.
Provided the voltage and current ratings are adequate for the location, ANY transistors will work anywhere.
I even tried 2N3055’s (Ft=10kHz) for the final output transistors. It didn’t quite work. But anything with an Ft > 50kHz will work and most modern power transistors have much larger Fts.

You can open up the schematic in another tab in you browser. On a PC, right-click you mouse over the image. If using a tablet, then keep your finger pressed on the image for a second or so until the option comes up.

Thanks Dermot ... I fully agree! ... Long time user of both vacuum tube class B amplifiers and Spice simulation (I started with the Berkeley version in the 70s), I would be very interested with your own LTSpice files ... Thanks in advance.

Very interesting and reminded me of my student days in the 60’s building the very similar Bailey and the Lindsley-Hood amplifiers. 1964 Wireless World I think. I believe that the greatest developments in HiFi since those days have been in the field of marketing

The Blomley amplifier is hugely different in principle to the Bailey or Linsley-Hood amplifiers (very very good though they are). It separates the processes of rectifying the signal and amplifying into two separate places in the circuit. All distortions without any feedback at all are below 1%. (put a 1F (! because you can in a simulation) capacitor to earth in each of the feedback paths.

I agree about the marketing. You have to LISTEN carefully and it becomes obvious…

Good evening,
I would like to know how good can be any amplifier with a single Voltage suply ? In my opinion it can only be as good as the capacitor itself: this is a weak point indeed ! anything excellent ahead of a capacitor will always be muffled by its impedance and irregularities.
Do you have by any chance the same kind of study of this amp with a split supply, say +/-30 V ?

Billy60300 - If you follow the current path of the output stage fed from a split supply, through the speaker and back THROUGH the power supply to the point you started at, you will find that the AC signal does indeed go through a capacitor.

I don’t have a decent layout and would very much like one if you can. I think we would all appreciate it.

I will look at the transistors, but the ones I can get in South Africa will be very different to the much greater range that you can get in the Northern Hemisphere.

Basically any modern trasnsistor with a Vce and Vcb of greater than 80v and current of 100mA for the small-signal transistors. The two driver transistors TR9 and TR10 need to have a 1A capability. And the splitter transistors TR4 and TR5 should be RF transistors with an Ft > 100MHz. Final transistors (TR11 and TR12) need to have a 10A capability.

So - I couldn’t resist and made a start before seeing your reply.
Went back to Blomley’s schematic to find that I still have some of the original BCnnn transistors Have a schematic, working on layout, will keep you posted. I’m using through hole components so that will be easier to try different devices.

Thanks. I started laying out a PCB but quickly had to shift gears to a contract job. I look forward to seeing what you come up with. Although SMD is so much easier from a manufacturing standpoint and the larger SMD’s are do-able for hobbyists, I use through hole for most things I design/build if they are just for me. It certainly makes experimenting with different components for audio and even (some) RF applications much easier for a hobbyist without access to microscopes or reflow ovens.

I’m trying to simulate it and it keeps clipping with anything greater than about 0.25V 1kHz sine wave at the input. I’m guessing this isn’t intended for line level audio? Or maybe something’s wrong in my simulation. Either way I’d like to build one that can handle pro line level audio.

Thank you for a great article! I teach Class B amplifiers at the very end of our Electronics 2 class and will include this in my student’s reading material. I would like to have the LTSpice file if possible. Again thanks!

I’ve never built an audio amplifier, but I think this could make a rewarding first attempt. Since you’ve successfully built these in the past, would you consider a follow-up article with a modern parts list and an enlargeable schematic?

Also, one question: Could you explain what you mean by, “Great care must be taken to physically separate the input from the output…”? Are you saying they should be at opposite ends of the PCB, or separate PCBs (if this, where would you divide the circuit?), or something else?

Being inherently a very high frequency amplifier (works at RF if you want) the input and wires must be physically far away. Don’t need to build onto separate boards - just be careful with routing. My first amplifier had a single edge-connector for the board with all connections coming in there - not a good idea.

Nice article, and I concur with the sensitivity of the ear. I have been in R&D, in audio, for some 40 years (50 years total), and found that alterations in the frequency response (fr), with accompanying changes in harmonic and IMD amplitude pose a real problem.

In one of my experiments,I have incorporated a 9k resistor across a driver. Changing a 9k ohm resistor by two hundredths of an ohm very slightly varies the Fr, and thus the harmonics and IMD products. I say slightly due to the fact that I am dealing with 1 part in ~450,000. It could be less. That is an incredibly small change in fr and individual harmonics, and IMD products. Thus +/- 0,1 db from 20-20khz means nothing. By the way, Olson’s research concluded, which I concur, that the higher the harmonic number, the more sensitive the ear is, at least under some conditions.

However, as mentioned below by another poster, the design will most likely never be totally accurate due to parts quality, in the power supply and around the active parts. This is especially true with power supply filter capacitors, the last being foremost of the caps in the signal circuitry.

The power supply filter capacitors and especially the last filter capacitor, has problems with ESR, DA, termination techniques etc. due to its large size. Here is a link/article, written by Walter Jung and Richard Marsh, that gives some general idea of the problems.

Tube circuits also have problems as well, but smaller filter capacitors does lessen the ESR and DA problems. With that said, using specialized listening tests has confirmed that PP AB1 amps can have perfect sound, except possibly near the 20 hz level, where the bass could be slightly less. However, as the article mentions, SS virtually never passes the test in accuracy.

cheers

Steve Sammet (retired)
SAS Audio Labshttp://www.sasaudiolabs.com/pream11a.htm
ps. The pages are up for those who wish to gain some information on used components that might come up for sale (Which virtually never does). Nothing is being manufactured, retirement.

A very well written article, thank you. I only woke up a couple of hours ago, and after reading it I’ve already learnt a few new things. The idea of presence is a compelling one, and I’m sure many of us now are itching to build it and have a listen. I can only imagine was an incredible guitar amplifier it would make too.

Very good article, Dermot! I am interested in building this circuit, could you please send me the Spice files? Thank you!
One other question… which is the best way to power this amp for a power supply? (From 110V 60Hz to ...?)

I wish I could simply bypass the inaccurate information provided, but the designer’s article has some problems that need addressing.

1) The rise time/fall time of the amplifier is Not the only rise time we have to deal with in the audio system. We have the source, preamplifier, and speakers as well. As such, the audio system’s rise time/fall time becomes much longer than just the amplifier’s value. For simplicity, if the frequency response of the amp is -1db at 25khz, and the preamplifier’s is -1db at 25khz, the fr of the two components is -2db at 25khz, and -1db in the teens khz. As such, we have altered the rise time of the system, which is perceivable.

Two studies demonstrated that the ear perceives rise time changes of at least 5 microseconds (us) 66.6khz, Jneutron (FermiLab, CERN, Brookhaven National Laboratory PhD EE) claims to have read a study of 2us, 166,66khz. I personally have altered the -1db from 200khz to 150khz back in the 1980s, individuals have perceived the sonic difference.

2) Next, to keep things simple, we have the problems of ESR and DA.
A. ESR is a series of resistances with intervals of capacitance between each resistor (theoretically infinite), thus the ESR can Not be linear.
B. The DA voltage of a capacitor varies over minutes, not microseconds. As such, it can vary the bias of an active device, depending upon the impedances involved, and insulating material used. Electrolytics are normally in the vicinity of 4% or so DA.

3) Next, the fr influences not only the overall frequency response, but the amplitude relationship between the fundamental and harmonics. The ear is extremely sensitive to such, the higher the harmonic the more sensitive the ear.

4) One could not perceive a sonic difference between capacitors, obvious either the audio system is quite inferior. By the way, Walter Jung’s article clearly demonstrates how inaccurate large electrolytic capacitors, used in the power supply, are. The last one or two filter caps are clearly in the signal path. The output electrolytic capacitor is so so, but not exceptional.

5) Did I miss any mention of layout to prevent channel separation problems? It is important as in a typical home setup, the sonics will not mimic the lab measurements.

I don’t mean to be so ruff, and one can build the amp, but please don’t expect totally accurate performance, because it won’t.

Great article! It is inspiring me to build the circuit and test it. Let us know when you upload the LTSpice files. Also, a recommended BOM including some suggested alternate components would be good. Some of the Transistors listed in the circuit diagram are obsolete and are tricky to source.

An interesting article, and I’ve long thought intermodulation distortion, not harmonic distortion is what causes a muddy and indistinct sound. But I don’t think you can have one without the other as they’re both produced by any non-linearity.The classic example is the extreme non-linearity of a diode used to demodulate an AM signal - all it’s doing is creating sum and difference intermodulation products between the carrier and sidebands, which, of course, include the audio signal. So if an amplifier truly has negligible harmonic distortion it can’t produce intermodulation distortion.

Also, I’m not quite convinced that the time the feedback signal takes to get back to the input can be a factor with transistors having an Ft orders of magnitude beyond the audio. I confess that my grasp of poles and zeros in my graduate control systems course many years ago was less than complete, but any significant delay would surely cause instability.

But I’ll stop short of dismissing it all as snake oil. Lab tests of THD using a continuous sine wave from a signal generator are unlikely to reflect real world performance. Transients may cause the the power supply to dip due, for example, to capacitor ESR. And when did you ever see a THD figure for those horrible analoggy things that covert your finely honed electrical energy into sound? In fact, capacitors, too, are non-linear, which is why the dream of supercapacitors replacing batteries has failed to materialise. (At high electric field strengths the energy stored by a dielectric no longer increases with the square of the voltage.) So I suspect the debate will go on.

Nice Article!
I created the LTspice simulation from the picture in the article. Ithink there is an error in R22 and R23 with the value of 1m. LTspice will interpret this as 1 milliohm.
For soft clipping maybe 1 meg would be better? Then “1meg” should be used.
Using 0 milliohm as default ESR in caps and power-supplies lead to overoptimistic simulations. Insert some realistic number here.

Power supply caps ESR plus the source impedance of the power supply generates a load dependent voltage that will find its way the the loudspeakers. This is seldom discussed. Try inserting this in your simulation. In real life you could load the amp with a power resistor and tap the signal in the power supply, maybe to another amp and listen to, or measure, the junk from the power supply.

The audiophile area is a minefield of facts, opinions, and emotions. It is often hard to sort thru what you hear and read. Much of the discussions of amps etc are trying to separate facts from emotions and with marketing efforts included, to make life easy. Professional musicians one should think have the best taste in selecting sound reproducing equipment but the reality is different.
My opinion is that musicians listens to music and audiophiles listen to what is wrong with the music.

I think making a little PCB for this amp would be a snap. Blomley had some real balls when he said the amp was too good for the common people.

You can measure all kinds of things in audio signals, but no instrumentation exists that I know could find out if a music piece was played badly, which the ear could find instantly. Sound reproduction equipment are measured and quality graded with instruments that are available, regardless of relevance, with criteria carried over from the vacuum-tube era.
Strangely enough the old tube stuff sounded good compared with early transistor amps. They measured horribly compared to solid state amps but still sounded better. The challenges with tube amps have now been easier to solve. Power supplies can be made much more robust and quiet. DC-powering filaments is no problem either to help getting rid of hum.
Making a good tube amp will be expensive, output transformers and the tubes are not cheap. The reward could be good. Tubes have a much simpler transfer function than bipolar transistors and simpler to explain.
Either way you will have fun.

I ran some simulation with LTspice and found out some things.
The 5KHz simulated signal source is rife with harmonics. Surprise! Stronger odds than evens.
The soft clipper works OK with anything from 1Meg to 640K @ R22,23. Even order harmonics are generated to match the odd ones when approaching saturation.
Removing the R6 feedback resistor lowers harmonics 20dB, and matches the input source.
Power supply rejection is -55dB to the output.
I designed a 10x10cm PCB a Sunday afternoon if anyone is interested in stuffing it. I found some interesting 100V 15A 4MHz Darlingtons for the output. I added an extra set of these for more current capability. Ballast resistors may be needed for even current sharing, perhaps, but part tolerances and using the same heatsink may help the match.
Other fatter complementary output transistors are available in Darlington config with 120V 25A 50MHz for a few bucks more.
This circuit lack any kind of protection, like fuses, inductor in the output. Maybe not needed?
The input is not symmetric like later amps, and overloads asymmetrically. Upper TR7 base drive is 17 times lower than the TR8 bottom half.
Lots of stuff to look at.

Interesting simulation results. I have also made my own LTspice model based upon the picture in the article. Note that there are also a few type-o errors which could cause some confusion (e.g. R9 should be 6.8k, R3 should be 4.7k). I tried 1m, 1k and 1Meg for R22 and R23. I checked the picture in this article against the original 1971 circuit diagram and note the two added transistors TRP1 and TRP2. Think I modelled the circuit correctly but there seems to be an error in my model cannot get stable output signal. I would like to compare LTSpice models. There is a LTSpice Yahoo Group. Is anyone interested in collaborating there on this LTSpice model?

“...Note that there are also a few type-o errors which could cause some confusion (e.g. R9 should be 6.8k, R3 should be 4.7k)...”
Not a typo per se, but an alternative way of expressing a decimal point, this is commonly used as a dot may disappear in small print.

The “..RP…” labeled parts belong to the soft clipping circuit not included in the 1971 Peter Blomley schematic.

My simulation runs without any problems, look for errors.
This is not not a “model” but a simulation schematic. “Models” are the individual part used in common lingo.

Yes. join the Yahoo group. I have an account there. They have loads of models you cannot find elsewhere.

The sim will run much faster if you skip the initial DC operating point. The startup includes a spike that will pummel you eardrums. Back in the day relays where used for outputs. Also temperature compensation diodes etc. I sorta lost interest in Transistor amps as the tube amps sounded better. This could be a fun little circuit to play with.

Another transistor amp I heard in the early 70’s was the Spectra Sonics 701, which sounded really good and also had an excellent reputation in pro circles. It has some similarities with this and is a more classical OP amp differential design. It required the use of one input audio transformer per channel, but it could easily be hooked up in a bridged config, bi-amping and such could be easily done. PNP power transistors were not so good back then and that amp used a quasi-complementary output, with six TO-220 Darlingtons, which you still can get.
Those amps where on plug in cards and gobs of them could be stuffed in a chassis.

Thanks for the tips and lingo corrections. I checked my simulation schematic and there were no schematic errors. As you suggested I deleted the startup input spike (i.e. changed “.tran 0 1 0 startup” to “.tran 0 1 0” and deleted “.include relay.lib” and it worked! The circuit amplified the input sine wave +- 3mV to +-300mV. Thinking I can now use this model to figure out each resistor power ratings needed. Any tips on choosing best resistors or tips on what small signal diode can be used to replace the Germanium diode (i.e. 1N34A)?

If you shoot me the files, I’ll throw a couple together and measure the performance with my (admittedly) pedestrian (HP/Agilent/Rigol/Instrument and Measurement mics) gear. I started laying out a PCB but had to shift gears to a more time sensitive project.